Re: RE: Re: [asterisk-users] setting call-limits

Hi, I believe you need to setup hints for call-limit to work. can you explain? I don't find any information on it... is this a tool or a library? Thanks -- Feel free – 10 GB Mailbox, 100 FreeSMS/Monat ... Jetzt GMX TopMail testen: http://www.gmx.net/de/go/topmail

Re: [asterisk-users] Please suggest me Best VoIP Service Provider

Hi Gbenga Great,I want to make calls to USA from INDIA using our "Asterisk". The sound quality should be good and able to make 4 Simulatanious calls at a time to USA. Do u have any unlimited plan to make calls to USA from INDIA using Asterisk? Please provide full details of your service. Looking

[asterisk-users] Solution init.d scripts for CentOS 4.3

Hi Everyone, I was having a lot of trouble starting up Asterisk and zaptel using the init.d scripts. I have worked on the scripts and now the zaptel script so it reads preferences of /etc/sysconfig/zaptel file and starts the zap interfaces properly. The asterisk init.d script does not load or

RE: RE: Re: [asterisk-users] setting call-limits

can you explain? I don't find any information on it... is this a tool or a library? Look at http://www.voip-info.org/wiki-Asterisk+standard+extensions There's an example on how to setup hints under standard priorities. ___ --Bandwidth and Colocation

RE : [asterisk-users] Asterisk and H.323

Hello, Try both chan_oh323 and gnugk . Harry --- Aaron Anderson [EMAIL PROTECTED] a écrit : I have been scouring the net the last couple of days looking for some kind of tutorial or walkthrough on setting up a h.323 channel in asterisk. What I need to do is basically this: I have a

[asterisk-users] IAX2 trunking problems

Hello list We are having some strange problems. When we setup trunking between two of our servers, the connection only uses trunking one way. Ex: Data From callingserver to receivingserver uses trunking Data from receivingserver to callingserver does not use trunking. I discovered this

RE: [asterisk-users] Asterisk autoloading of card modules

My /etc/sysconfig/zaptel configuration has only one MODULES directive enabled MODULES=$MODULES wctdm However when I start asterisk it loads the wct1xxp module. Which configuration file controls the loading of card modules? Check /etc/modprobe.conf I clear that out and just leave the module

[asterisk-users] NAT

I am interested to configure my linux box for a server for my call center. What sort of NAT/IPTABLES I need to implement on my server? I have just masqurate the box and it is not workingbut i can have local calls .. Thanks in advance. atif ___

Re: [asterisk-users] Please suggest me Best VoIP Service Provider

Hi we are located in hyderabad (india) where are you located ? we do support DID incoming and out going we have veryresonable rates for USA and other Countries contact me with your Phone ram On 7/24/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi Gbenga Great,I want to make calls to USA from

[asterisk-users] overlapdial and DID not always working

Hi, another try with a hopefully better subject. I am here in Germany connected to the telephone system with a PRI interface: 00:0b.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface. I am using asterisk 1.2.7.1 and zaptel 1.2.5. To let DID work, I have set the options

Re: [asterisk-users] Missing close quote in CallerID breaks SIP. . .workaround?

Brian Capouch wrote: I posted about this some while back, and at that point was told the remote end is broken, nothing we can do about it. The problem: for whatever reason, some CallerID names come in broken. There is an example CLI trace shown below. My question: is there anything I can

[asterisk-users] Asterisk Jobs.com Updates

Greetings Asterisk Job Seekers, Asterisk Jobs is proud to announce reaching over 100 users. We are actively marketing the site to potential employers. More jobs will be posted on the site as more users join and traffic expands. We will be introducing new paid plans for employers after our

[asterisk-users] IP CDR

Hi Please how can I get the user register ip address and put it at cdr ,its too important Thanks * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail

[asterisk-users] Conference

Pleasehow can I enable the 3way-conference on a sip gateway like (addpac) from dialing an extension since I the gateway do not have this feature . Regards * No employee or agent is authorized to conclude any binding agreement on

Re: [asterisk-users] overlapdial and DID not always working

Sebastian Reitenbach ha scritto: but when I issue a reload chan_zap in the asterisk console, then I can see the following in the log output: Jul 21 14:20:16 WARNING[1345] chan_zap.c: Ignoring signalling Jul 21 14:20:16 WARNING[1345] chan_zap.c: Ignoring switchtype Jul 21 14:20:16

[asterisk-users] Transfering Problem

Hi guys, i want to know why call transfering doesnt work with queues. i have passed Tt to the Queue() application. when i press #, asterisk plays pbx_transfer followed by dialtone. after dialing the extension nothing happens. I have tried to transfer the call without queues with the help of

Re: [asterisk-users] overlapdial and DID not always working

Sebastian Reitenbach ha scritto: so about 80% of the incoming calls work well, but especially with one sender we have a problem, there is always the last digit missing. This is a 1-800 service in the US, forwarding the call to our asterisk. As a workaround I configured it to call to

Re: RE : [asterisk-users] Asterisk and H.323

I have been messing with both all day. I think what might be tripping me up is the extensions.conf. I was able to receive an incoming connection from the client, but all the system returned was a busy signal. This call was to a known good number (my phone) so I'm not sure what's wrong. Will

Re: RE : [asterisk-users] Asterisk and H.323

On 7/24/06, Aaron Anderson [EMAIL PROTECTED] wrote: I have been messing with both all day. I think what might be tripping me up is the extensions.conf. i do think so too :) I was able to receive an incoming connection from the client, but all the system returned was a busy signal. This

[asterisk-users] Multiuser and analog port

Hi,A couple of weeks ago, I've read in this list that some high end wireless analog phone systems provided some sort of DID features.I can't put a hand on this thread.Does anyone have a clue ?What I'm after is : Analog wireless phone systems connect several handsets to a single analog line.For

[asterisk-users] Regular expression problem

Hi! What's wrong with this? exten = s,1,Set(myvar=nothing) exten = s,2,Set(myvar = $[${CALLERID(num)} : ([a-z]+)]) exten = s,3,NoOp(${myvar}) The regular expression in priority 2 matches, but the result is not assigned to variable myvar, on the console, I see this: -- Executing

[asterisk-users] Mitel 3300 + *

Has anybody managed to get * working with a Mitel 3300 and if so what method (sip / h323) and licences did you use and have you any tips or pitfalls. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

Re: RE : [asterisk-users] Asterisk and H.323

Thanks for the response. I have been able to now receive calls over h.323 using sjphone through the built in ooh323 channel driver. It seems to work ok for a bit but then asterisk seems to stop accepting connections and the server needs to be rebooted. On a slightly different note, I have 3

Re: [asterisk-users] overlapdial and DID not always working

Sebastian Reitenbach wrote: any idea what I can do? especially why it says it ignores the overlapdial parameter, and why it is accepting them nevertheless? are there any timing parameters to tell asterisk to wait a second longer for the last digit? some rx.. tx.. parameters in the zapata.conf?

Re: [asterisk-users] Regular expression problem

You are using quotes when you should not be. Notice the double quoting of -- Executing NoOp(SIP/n-5d23, nothing) in new stack Benjamin Stocker wrote: Hi! What's wrong with this? exten = s,1,Set(myvar=nothing) exten = s,2,Set(myvar = $[${CALLERID(num)} : ([a-z]+)]) exten =

Re: [asterisk-users] Voicemail not sent via email

Dean @ INKnBITs wrote: I have setup the voicemail.conf as below, but I not receiving any emails. Any thoughts? voicemail.conf [default] 3002 = 1234,Bob Wright,[EMAIL PROTECTED],,|attach=yes I have also uncommitted the mailcmd=usr/sbin/sendmail -t but that does not work. Check the logs on

Re: RE : [asterisk-users] Asterisk and H.323

On 7/24/06, Aaron Anderson [EMAIL PROTECTED] wrote: Thanks for the response. I have been able to now receive calls over h.323 using sjphone through the built in ooh323 channel driver. It seems to work ok for a bit but then asterisk seems to stop accepting connections and the server needs to be

Re: [asterisk-users] Voicemail not sent via email

2006/7/24, Dean @ INKnBITs [EMAIL PROTECTED]: I have setup the voicemail.conf as below, but I not receiving any emails. Any thoughts? voicemail.conf [default] 3002 = 1234,Bob Wright,[EMAIL PROTECTED],,|attach=yes I have also uncommitted the mailcmd=usr/sbin/sendmail -t but that does not

RE: [asterisk-users] Voicemail not sent via email

Make sure your mail system is working. Try mail [EMAIL PROTECTED] From the os command prompt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean @ INKnBITs Sent: Monday, July 24, 2006 8:02 AM To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Regular expression problem

2006/7/24, Eric ManxPower Wieling [EMAIL PROTECTED]: You are using quotes when you should not be. Notice the double quoting of -- Executing NoOp(SIP/n-5d23, nothing) in new stack Thanks for reply. I removed the quotes, but it did not solve the problem. After lots of trials I've found the

Re: [asterisk-users] Regular expression problem

On 7/24/06, Benjamin Stocker [EMAIL PROTECTED] wrote: Hi! What's wrong with this? exten = s,1,Set(myvar=nothing) exten = s,2,Set(myvar = $[${CALLERID(num)} : ([a-z]+)]) exten = s,3,NoOp(${myvar}) Try removing the spaces on either side of the = symbol. Regards, Gonzalo

Re: [asterisk-users] Asterisk autoloading of card modules

Hi Alejandro, Thanks for your suggestions. Where did you fetch your rpms? I had to fix up the init scripts for everything to work On 7/24/06, Alejandro Kauffmann [EMAIL PROTECTED] wrote: My /etc/sysconfig/zaptel configuration has only one MODULES directive enabled MODULES=$MODULES wctdm

[asterisk-users] Connecting Asterisk to a Metaswitch

I am having a difficult time connecting an Asterisk box to a Metaswitch. I looked at the page at http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+to+Metaswitch but was not able to make much progress. If someone could direct in what direction to start troubleshooting this problem I

Re: [asterisk-users] Voicemail not sent via email

Sounds very much like a sendmail issue - check /var/log/maillog PaulH On Mon, 2006-07-24 at 13:01 +0100, Dean @ INKnBITs wrote: I have setup the voicemail.conf as below, but I not receiving any emails. Any thoughts? voicemail.conf [default] 3002 = 1234,Bob Wright,[EMAIL

[asterisk-users] Conference

Pleasehow can I enable the 3way-conference on a sip gateway like (addpac) by dialing an extension since the gateway do not have this feature ,I want to make it on server level or if you know how conference work . Regards * No employee

Re: [asterisk-users] NAT

VoIP-Info.org has some information regarding this:See here: http://www.voip-info.org/wiki-NAT+and+VOIPAnd here: http://www.voip-info.org/wiki/view/Asterisk+firewall+rulesAlexOn 7/24/06, Atif Munir [EMAIL PROTECTED] wrote:I am interested to configure my linux box for a server for my call center.

RE: [asterisk-users] Connecting Asterisk to a Metaswitch

I recently got this going, and had a similar experience. In my case, the solution was to set the RPID to the expected number assigned to the account. YMMV, but it's worth a try. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kharris

[asterisk-users] Asterisk and Phonesystems ...

Hi on a new Asterisk installation, i have a small problems with Asterisk and the VoIP Operator PhoneSystems. Anyone have connected Asterisk to Phonesystems ? I have this when i want call: chan_sip.c:9696 handle_response_invite: Forbidden - wrong password on authentication for INVITE to

[asterisk-users] compain

I have configured Asterisk and can have calls from one ext to another.But from where i can get the call center compains to make calls for my call center setup? and where/how that sort of stuff will be placed to get the info for the agent? Thanks in advance. atif

Re: [asterisk-users] PRI got event: HDLC Abort (6) on Primary D-channel of span 1 (fwd)

i had the same Problem, but i don't know why, i spitted the second card in a second Asterisk Box, and its wokring, with 2 TE405P in one Server its not working. If you solved the Problem, please let me know. Thanks Nico On Thu, 20 Jul 2006, asterisk wrote: Hi all, We have two server

[asterisk-users] asterisk extra sounds: what for?

Hi, what are asterisk extra sounds (asterisk-sounds.x.x.x.tar.gz file) for?? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Please suggest me Best VoIP Service Provider

Hi Ram,I am also located in Hyderabad only. Please give me your website. I will go through your website. Looking forward to your response. Thank you.Regards,Chandra.ram [EMAIL PROTECTED] wrote: Hi we are located in hyderabad (india) where are you located ? we do support DID incoming and out

[asterisk-users] H.323 an IAX

Hi I have a problem with the NAT using H.323 and am thinking of employing IAX as a workwround. I have a scenario in my mind which I am not sure is gonna workor not, neaways here it goes. I want myIAX clients to connect to Asterisk which will be interconnected with H.323 terminating gateways. Now

RE: [asterisk-users] Voicemail not sent via email

I'm getting: 354+Enter+message,+ending+with+.+on+a+line+by+itself 0 0 54 0 4437 SMTP - - - - 550+Administrative+prohibition 0 0 30 0 4781 SMTP - - - - Any ideas? Thanks, Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric ManxPower Wieling Sent:

[asterisk-users] Asterisk, IAXModem and Hylafax

Hi all, I've got IAXModem setup and configured with Asterisk, things look OK here. I have Hylafax installed and I'm currently reading through he configs. I'm having a difficult time with Hylafax though, not understanding the user and mail settings very well, just fumbling around. Trying to

[asterisk-users] How to receive a phone call each time you receive an email ?

Hi,Which is the simplest way to be notified on your phone each time a given emailbox receives an incoming email ?I'm already running an Asterisk server and I'm wondering which mail server would do the job. Ultimately, I would try to tune the mail server to recall as long as the incoming mail is

RE: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk

With pen in hand, Douglas Garstang succussfully stormed bulwarks which others armed with sword and excommunication have been repulsed, and said ... Well in that case, what's the point in having the ATA register with Asterisk? You just direct all PSTN-VOIP calls to Asterisk with their PSTN CID

RE: [asterisk-users] How to receive a phone call each time you receivean email ?

Take it back to the old-skool! Use biff..or a newer version ebiff, Yamb, etc. etc. Snip ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] H.323 an IAX

On 24 Jul 2006, at 14:24, Asif Ali wrote: Hi I have a problem with the NAT using H.323 and am thinking of employing IAX as a workwround. I have a scenario in my mind which I am not sure is gonna work or not, neaways here it goes. I want my IAX clients to connect to Asterisk which will be

RE: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk

Sheesh! forgot to give the url http://nerdvittles.com/index.php?p=65 JC With pen in hand, [EMAIL PROTECTED] succussfully stormed bulwarks which others armed with sword and excommunication have been repulsed, and said ... With pen in hand, Douglas Garstang succussfully stormed bulwarks

Re: [asterisk-users] How to receive a phone call each time you receive an email ?

There must be several ways, however one that comes to my mind is use dnotify http://oskarsapps.mine.nu/dnotify.html and execute a command that create a .call file everytime a new file is created in the mailboxes http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out To call the

RE: [asterisk-users] Mitel 3300 + *

The 3300 uses the MiNet protocol so you wouldn't be able to interface it over a LAN, but hooking it up the "old school" way (crossover PRI cable) should work fine, you would have to set up DID's on your 3300 that correspond to the extension numbers you would want to dial the 3300 from

[asterisk-users] playback / stream file

Hi all, I wonder if there is a simple solution for my needs: I have a archive, that includes different soundfiles in mp3 format. Some 11khz some 44.1khz and so on. Now I want to play them back to a caller, but asterisk only can play them in mono 8khz. So is there a possibility to use any

[asterisk-users] Astrisks compatable cards

Hi I have visited pbxeq.com they are selling below cards casy Asterisk compatable A400P04 A400P40 A1200P01 how best this product.. any one in this group used compare to digium cards, how best these cards Looks price is reasonable, but before iam buying them want to feed back just trying to

[asterisk-users] AstLinux 0.4.2 Released

Hello everyone, AstLinux 0.4.2 has been released. Updates include: - Asterisk 1.2.10 - Zaptel 1.2.7 - libpri 1.2.3 - codec_g729a The Digium g729a codec has been included in this release. The codec has not been available since AstLinux 0.4.0 when the C library was switched to

[asterisk-users] ZapRAS

Hi all, I'm trying the ZapRAS application and I followed this giude. Could it work with an HFC pci isdn card? I tryied and i get this error: *CLI -- Executing Answer("Zap/1-1", "") in new stack -- Accepting data call from '123456789' to '987654321' on channel 0/1, span 1 -- Executing

[asterisk-users] reboots itlself

I have an AAH, seems to be Asterisk version 1.2.7.1. It seems to be rebooting everyday around 8:30 am and the office goes hay wire, as this is a doctor's office, even if it's for a brief minute. Nothing remarkable in the logs. Please help ? -balu raman Ryder Brook PediatricsP.O.Box

[asterisk-users] Operator in Voicemail

Ive got an odd problem. I have set in Voicemail.conf operator=yes as a default. This is so that when a caller is in the voicemail system they can press 0 and be sent to the operator. This works fine when the caller is internal to the system but NOT when the caller is calling in from the

[asterisk-users] Transfers - No ringback or moh

I don't know why, but when doing transfers between Polycom phones, once the transferring party hits transfer a second time, to be removed from the call, User A no longer hears music on hold, or a ring back. Scenario. 1. User A dials User B. 2. User A and User B are connected. 3. User B

[asterisk-users] FW: Conference

Pleasehow can I enable the 3way-conference on a sip gateway like (addpac) by dialing an extension since the gateway do not have this feature ,I want to make it on server level or if you know how conference work . Regards * No

RE: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk

-Original Message- From: Jonathan Attwood [mailto:[EMAIL PROTECTED] Sent: Saturday, July 22, 2006 12:45 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk For the OP, do

Re: [asterisk-users] overlapdial and DID not always working

Hi, This is normal, some parameter cannot be changed with reload, the only way to change them is stop asterisk and restart (stop now, restart asterisk, you found all parameters correct). This message is like ouch you are reloading the configuration, but i cannot change this parameter by

Re: [asterisk-users] overlapdial and DID not always working

Hi, we have a problem, there is always the last digit missing. This is a 1-800 service in the US, forwarding the call to our asterisk. As a workaround I configured it to call to X580 and have an inbound route set for X58 to the number I want to reach. any idea what I can do?

[asterisk-users] Circuit/channel Congestion

Hello all. I have a Digium TE110P board and when I do a 'dial 4509' I got: Jul 24 11:44:33 NOTICE[8180]: app_dial.c:1049 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) My zaptel.conf: span=1,0,0,cas,hdb3,crc4 bchan=1-15,17-31 dchan=16 My

[asterisk-users] Asterisk and Vigortalk problem

Hi All on the list, I'm having a problem with some Draytek Vigortalk, firmware v2.5.8 registered to an Asterisk v1.2.9.1 (AAHv2.8 updated). Basically if I make a call from one of those Vigortalk extension to another whatever extension I don't hear anything while the called extension may do. If

Re: [asterisk-users] overlapdial and DID not always working

I found the same indentical problem, the trouble was the switchtipe, i am using national and i switched to unknown. is unknown allowed for switchtype? when I take a look here: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf then there is no unknown switchtype? jut

[asterisk-users] Clocking Multiple T1 Cards

[ I originally sent this to the list last week but it never arrived; it may have been stuck in moderation because the sending address is not my subscribed address. I apologize if you get this twice. ] I have a Digium TE205P connected to two channel banks in my Asterisk. PBX. I will be

RE: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk

Is it a Sipura 3000? If so you can use the link below, and if the ata is on the network, you can enter the IP address and it will setup the ata for you, and gives you the details to enter into asterisks. (If you use the bottom option) Only thing is you have to signup to use the wizard. Worked

[asterisk-users] Odd SIP timeout

I have two SIP providers that work fine during the day but it seems that during extended periods of non-use one provider gets stale and the refresh shows an odd number. A simple reload of SIP clears the problem but don't want to rely on cron as a solution. I qualify them both and I'm not behind

RE: [asterisk-users] Mitel 3300 + *

Thanks, I know the phones can be loaded with the sip protocol and seem to recall from some time ago that mitel said they were going to introduce sip to the platform and assumed they had. Oh well, back to the drawing board. neil From: [EMAIL PROTECTED] [mailto:[EMAIL

Re: [asterisk-users] Transfers - No ringback or moh

Douglas Garstang wrote: 1. User A dials User B. 2. User A and User B are connected. 3. User B hits the transfer soft key. User A gets music on hold. 4. User B dials user C. User C's phone rings, and user A continues to hear music on hold. 5. When User B presses the transfer soft key again to

Re: [asterisk-users] Circuit/channel Congestion

Lincoln Zuljewic Silva wrote: Hello all. I have a Digium TE110P board and when I do a 'dial 4509' I got: Jul 24 11:44:33 NOTICE[8180]: app_dial.c:1049 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) Where is your configuration for extension

Re: [asterisk-users] Clocking Multiple T1 Cards

Shaw Terwilliger wrote: [ I originally sent this to the list last week but it never arrived; it may have been stuck in moderation because the sending address is not my subscribed address. I apologize if you get this twice. ] I have a Digium TE205P connected to two channel banks in my

[asterisk-users] G729 Softphone

Looking for a SIP or IAX softphone for a call center application that can do G729 codec. Any recommendations? Ideally it would do screen pops, meaning that it will understand the URL option of the Dial command. Thanks, Daniel ___ --Bandwidth and

Re: [asterisk-users] Clocking Multiple T1 Cards

On Monday 24 July 2006 11:20, Shaw Terwilliger wrote: I have a Digium TE205P connected to two channel banks in my Asterisk. PBX. I will be installing a Sangoma A101 to be connected to a PSTN PRI in the same box. How should I go about configuring the T1 timing for these spans? Right now my

RE: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk

I am using an Sipura 3000 here and it is working (mostly) fine but I had a lot of learning to do in the beginning. I missed the original question and problem?? Perhaps you could state that again of refer me to it? Keep in mind there are two major firmware versions for the 3000. Version 2 and 3.

Re: [asterisk-users] Clocking Multiple T1 Cards

Shaw Terwilliger wrote: [ I originally sent this to the list last week but it never arrived; it may have been stuck in moderation because the sending address is not my subscribed address. I apologize if you get this twice. ] I have a Digium TE205P connected to two channel banks in my

Re: [asterisk-users] X100P clone not working

On Sun, Jul 23, 2006 at 02:21:01PM +0200, Frank Darner wrote: What is the output from 'cat /proc/zaptel/*' After delete of all Asterisk files and complete new install I got something: # modprobe zaptel modprobe wcfxo linux:/proc/zaptel # cat /proc/zaptel/* Span 1: WCFXO/0

Re: [asterisk-users] Circuit/channel Congestion

I have the same card, but in my zaptel.conf I have the following line span=1,1,0,hdb3,crc4 as you can see from the status your line is down. BR Thomas Lincoln Zuljewic

RE: [asterisk-users] Mitel 3300 + *

SIP is on the "low-end" product (3200?) which is meant as an Asterisk-style mini-PBX that provided an entry point to get Mitel into orgs with 50 users, but for some reason it seems discontinued I can't find it on their website anymore. There is a SIP stack for the 5220 and higher phones

Re: [asterisk-users] Circuit/channel Congestion

grep 4509 extensions.conf exten = 4509,1,Dial(Zap/g1/4509) Doug Lytle wrote: Lincoln Zuljewic Silva wrote: Hello all. I have a Digium TE110P board and when I do a 'dial 4509' I got: Jul 24 11:44:33 NOTICE[8180]: app_dial.c:1049 dial_exec_full: Unable to create channel of type 'Zap'

Re: [asterisk-users] G729 Softphone

On Mon, 2006-07-24 at 11:41 -0400, Daniel Salama wrote: Looking for a SIP or IAX softphone for a call center application that can do G729 codec. Any recommendations? Ideally it would do screen pops, meaning that it will understand the URL option of the Dial command. Give a try to

Re: [asterisk-users] Clocking Multiple T1 Cards

Andrew Kohlsmith wrote: Correct. Since you're using an entirely different card for the incoming PRI, you don't need to change these spans. Just add the third, and use '1' for timing to specify that you want the clock recovered from that span to be the primary clock source. As the other

RE: [asterisk-users] Operator in Voicemail

Im having the exact same problem here. I originally thought it was a context problem. However, to troubleshoot I tried placing the following in every context (default, from-inside, from-outside, etc) in extensions.conf with no luck: exten = o,1,DIAL(SIP/100,100) Like Kevin, it works

Re: [asterisk-users] Circuit/channel Congestion

Lincoln Zuljewic Silva wrote: grep 4509 extensions.conf exten = 4509,1,Dial(Zap/g1/4509) Doug Lytle wrote: Lincoln Zuljewic Silva wrote: Where is your configuration for extension 4509? I still don't know how you are hooking up extension 4509. Is this hanging off of a channel bank, is

Re: [asterisk-users] Circuit/channel Congestion

In my line, i need to use 0 because I send the clock to the other machine... Thomas Laurids Pedersen wrote: I have the same card, but in my zaptel.conf I have the following line span=1,1,0,hdb3,crc4 as you can see from the status your line is down. BR Thomas

Re: [asterisk-users] Clocking Multiple T1 Cards

On Monday 24 July 2006 12:11, Shaw Terwilliger wrote: Thank you; this is the kind of information I was looking for. The wiki and other documents told me exactly what the configuration options did, but I didn't know what kind of timing configuration was right for multiple cards. Essentially

Re: [asterisk-users] Cyberdata paging speakers - anyone use them?

For our stores, it would be nicer to have some kind of device that automatically mutes our music before playing input from the Asterisk pager. We already have a store full of speakers, no reason to duplicate them. Has anybody heard of such a thing? On 7/21/06, [EMAIL PROTECTED] [EMAIL

[asterisk-users] Intercom feature on Polycom phones

I have the situation where my client would like to Intercom an extension similar to auto-answer. I have polycom phones can this be done? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To

Re: [asterisk-users] G729 Softphone

I have the eyeBeam softphone but I don't see G729 in the list of available codecs (BTW, this is the paid version not X-Lite). Any clues? Thanks, Daniel On Jul 24, 2006, at 12:00 PM, Guillermo Salas M. wrote: On Mon, 2006-07-24 at 11:41 -0400, Daniel Salama wrote: Looking for a SIP or IAX

[asterisk-users] Asterisk Realtime Macros

Hi, I am trying to get asterisk Realtime to work. I have a fresh installed 1.2.10 setup on a debian system. I have taken the defaul setup and put it into the mysql database. I have setup two extensions 101 and 102. If I setup the extension like such: exten = 101,1,Dial(SIP/101) exten =

Re: [asterisk-users] G729 Softphone

Daniel Salama a écrit : Looking for a SIP or IAX softphone for a call center application that can do G729 codec. Any recommendations? Ideally it would do screen pops, meaning that it will understand the URL option of the Dial command. Of course I'm a little biased, but I think MozPhone is

Re: [asterisk-users] IP CDR

On 7/24/06, Khaled Chehab [EMAIL PROTECTED] wrote: Hi Please how can I get the user register ip address and put it at cdr ,its too important Thanks Check out this page: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sippeer I believe that some variation of that command and the

RE: [asterisk-users] Transfers - No ringback or moh

-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, July 24, 2006 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers - No ringback or moh Douglas Garstang wrote: 1. User A dials User B. 2. User A

Re: [asterisk-users] Intercom feature on Polycom phones

What do you mean by similar to auto answer? On 7/24/06, Stephen Murphy [EMAIL PROTECTED] wrote: I have the situation where my client would like to 'Intercom' an extension similar to auto-answer. I have polycom phones – can this be done? Steve

[asterisk-users] core dumps when phpagi script ends?

I've set up my IVR system as a PHP script using phpagi.php. The typical way for users to exit the system is by hanging up. This seems to work fine -- Asterisk promptly ends the call/connection -- but a side effect seems to be that it always leaves a core.pid file in

RE: [asterisk-users] Intercom feature on Polycom phones

Yes. http://www.voip-info.org/wiki/view/Polycom+auto-answer+config --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Murphy Sent:

RE: [asterisk-users] Operator in Voicemail

This is what I am using: exten = o,1,Answer() exten = o,2,GoTo(default,3000,1) exten = o,3,Hangup() Hope this helps, Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Davis Sent: maandag 24 juli 2006 18:20 To: Asterisk Users Mailing List -

Re: [asterisk-users] Circuit/channel Congestion

Its another pbx. Doug Lytle wrote: Lincoln Zuljewic Silva wrote: grep 4509 extensions.conf exten = 4509,1,Dial(Zap/g1/4509) Doug Lytle wrote: Lincoln Zuljewic Silva wrote: Where is your configuration for extension 4509? I still don't know how you are hooking up extension 4509. Is

RE: [asterisk-users] Operator in Voicemail

Hmm this works for me. I'm using 1.2.7.1, but doing VOIP only. No PSTN lines coming in. exten = o,1,Playback(walks-into-bar-mail) Currently i have a place holder for when my customer gets a real receptionist, then i'll substitute the Playback with a Dial application. this is placed in the

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